qt_rtmp_demo/media/AACAudioCoder.cpp

#include "AACAudioCoder.h"
//#include "Debuger.h"

using namespace AAC_CODER;
AACAudioCoder::~AACAudioCoder() {

}


void AACAudioCoder::OnAudioData(const void *frameaddress, uint32_t framelen)
{
	this->Encode((unsigned char *)frameaddress, framelen * 4);
}

AACAudioCoder::AACAudioCoder(unsigned int smprate, unsigned int channel) {
	AVCodecID codec_id = AV_CODEC_ID_AAC;

    pCodec = (AVCodec *)avcodec_find_encoder_by_name("libfdk_aac");
	if (!pCodec) {
		printf("Codec not found\n");
		this->mStatus = FAIL;
	}
	mCodecCtx = avcodec_alloc_context3(pCodec);
	if (!mCodecCtx) {
		printf("Could not allocate video codec context\n");
		this->mStatus = FAIL;
	}

	mCodecCtx->codec_id = pCodec->id;
	mCodecCtx->codec_type = AVMEDIA_TYPE_AUDIO;
	mCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;       ///< float
	mCodecCtx->sample_rate = 44100;
	mCodecCtx->channel_layout = AV_CH_LAYOUT_STEREO;
	mCodecCtx->channels = 2;
	mCodecCtx->bit_rate = 640000;
	mCodecCtx->time_base.den = 1;
	mCodecCtx->time_base.num = 23;
	mCodecCtx->frame_size = 1024;
	this->mObserver = nullptr;

	if (avcodec_open2(mCodecCtx, pCodec, NULL) < 0) {
		this->mStatus = FAIL;
	}
	mFrame = av_frame_alloc();
	mFrame->nb_samples = mCodecCtx->frame_size;
	mFrame->format = mCodecCtx->sample_fmt;
	int size = av_samples_get_buffer_size(NULL, mCodecCtx->channels, mCodecCtx->frame_size, mCodecCtx->sample_fmt, 1);
	mFrameBuf = (uint8_t *)av_malloc(size);
	avcodec_fill_audio_frame(mFrame, mCodecCtx->channels, mCodecCtx->sample_fmt, (const uint8_t*)mFrameBuf, size, 1);
	mPts = 0;
}

int adts_sample_rates[]={96000,882000,64000,48000,441000,32000,24000,22050,16000,12000,11025,8000,7350,0,0,0};

int FindAdstSRIndex(int samplerate)
{
	int i;
	for (i = 0; i < 16; i++)
	{
		if (samplerate == adts_sample_rates[i])
			return i;
	}
	return 16 - 1;
}
#define ADTS_HEAD_LEN 7

void MakeAdtsHeader(unsigned char *data, int samplerate, int channels, int iFrameLen)
{
	int profile = 2;  //AAC LC<4C><43>MediaCodecInfo.CodecProfileLevel.AACObjectLC;
	int freqIdx = 4;  //32K, <20><><EFBFBD><EFBFBD><EFBFBD><EFBFBD>ע<EFBFBD><D7A2>avpriv_mpeg4audio_sample_rates<65><73>32000<30><30>Ӧ<EFBFBD><D3A6><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>±꣬<C2B1><EAA3AC><EFBFBD><EFBFBD>ffmpegԴ<67><D4B4>
	int chanCfg = channels;  //<2F><><EFBFBD><EFBFBD><EFBFBD><EFBFBD>ע<EFBFBD><D7A2>channel_configuration<6F><6E>Stero˫<6F><CBAB><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>

					  /*int avpriv_mpeg4audio_sample_rates[] = {
					  96000, 88200, 64000, 48000, 44100, 32000,
					  24000, 22050, 16000, 12000, 11025, 8000, 7350
					  };
					  channel_configuration: <EFBFBD><EFBFBD>ʾ<EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>chanCfg
					  0: Defined in AOT Specifc Config
					  1: 1 channel: front-center
					  2: 2 channels: front-left, front-right
					  3: 3 channels: front-center, front-left, front-right
					  4: 4 channels: front-center, front-left, front-right, back-center
					  5: 5 channels: front-center, front-left, front-right, back-left, back-right
					  6: 6 channels: front-center, front-left, front-right, back-left, back-right, LFE-channel
					  7: 8 channels: front-center, front-left, front-right, side-left, side-right, back-left, back-right, LFE-channel
					  8-15: Reserved
					  */

					  // fill in ADTS data
    data[0] = (uint8_t)0xFF;
    data[1] = (uint8_t)0xF9;
    data[2] = (uint8_t)(((profile - 1) << 6) + (freqIdx << 2) + (chanCfg >> 2));
    data[3] = (uint8_t)(((chanCfg & 3) << 6) + (iFrameLen >> 11));
    data[4] = (uint8_t)((iFrameLen & 0x7FF) >> 3);
    data[5] = (uint8_t)(((iFrameLen & 7) << 5) + 0x1F);
    data[6] = (uint8_t)0xFC;
}

FILE *ptest = nullptr;
int once = 1;
int AACAudioCoder::Encode( unsigned  char *input, unsigned int num) {

	mFrame->nb_samples = mCodecCtx->frame_size;
	mFrame->format = mCodecCtx->sample_fmt;

	avcodec_fill_audio_frame(mFrame, mCodecCtx->channels,
		mCodecCtx->sample_fmt, input,
		num, 1);

	int aac_out_len = 0;
	unsigned char*aac_buf;
	if (nullptr == input) {
		return -1;
	}
	if (nullptr == ptest) {
		ptest = fopen("dst.aac", "wb");
	}
	av_init_packet(&pkt);
	pkt.data = NULL;    // packet data will be allocated by the encoder
	pkt.size = 0;

	int got_output = 0;
	mFrame->pts = mPts += 23;
	int ret = avcodec_encode_audio2(mCodecCtx, &pkt, mFrame, &got_output);
	if (ret < 0) {
		printf("Error encoding frame\n");
		return -1;
	}
	if (got_output) {
		if (nullptr != mObserver) {
			mObserver->OnAudioEncode(pkt.data, pkt.size, mFrame->pts);
		}
		fwrite(pkt.data, 1, pkt.size, ptest);
		av_free_packet(&pkt);
	}
	return 0;
}

int AAC_CODER::AACAudioCoder::SetObserver(EncodeAudioObserver *p)
{
	if (nullptr == this->mObserver) {
		this->mObserver = p;
	}
	return 0;
}